I'm not really good at voice - so please bare with me :)
I have a situation where I cant make a voip call via SIP using class4/5 softswitch behind NAT/PAT network.
The diagram :
NAT/PAT --- cloud/MPLS --- softswitch.
the softswitch provides IP centrex service - so there will be caller-group. the 2nd problem was that in a caller-group It cant establish a call origin from ip 1.1 back to ip 1.1. And i cant touch that softswitch (its xener - i dont exactly know what type). I'm wondering this softswitch capability - anyone using it?.
We have tested using other SIP server (using asterisk-based softswitch) and sniffed all SIP-related traffic - we have 403 error and the like - but my opinion its the PEs NAT router that dropped the SIP handshake - so the RTP wont pass-thru both caller/called party.
Modifying a single PE probably easy - but my catch is that - as long as I have some NAT router/firewall along the PE and softswitch path it will not work, correct?
Before i go further with Cisco Unified Border Element and Session Border Controller proposal - anyone would like to give me a comment about my understanding from above scenario?
any help would be appreciated,